Hi all

After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive

so here i have done some good developements

for testing iam doing all in one Server

Step1 :

Installed in Fresh BOX with Debian

Asterisk and A2B working Fine


Step2 : registered with SIP account iam able to make calls successfully

Step3 :

installed Opensips

Made Subscribers to view from A2b Database

Step4 : changed Asterisk port from 5060 to 5062

Step5 : Opensip config made changes to register users with Opensips
and when they dial 001X call send to Asterisk box


route[3]{

if (uri =~ "sip:001[0...@*"){
log(1, "Forwarding to Asterisk \n");
rewritehostport("A2b-asterisk-IP:5062");
route(1);
exit;
}

Works Fine, No problems as of now

But to go in advance, i want to use Number of * boxes to achive more Load

Step5 : added Dispatcher Module in the Opensips

loadmodule "dispatcher.so"
.
.
.
modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg")
.
.
.
.
changed route to use dispatcher

route[3]{

if (uri =~ "sip:001[0...@*"){
log(1, "Forwarding to Asterisk \n");
ds_select_dst("2","4");
forward();
route(1);
exit;
}


My dispatcher Config Looks like below

dispatcher.cfg
2 sip:a2b-asterisk-ip:5062
2 sip:a2b-asterisk-ip2:5062

I have restarted Opensips

when i dial 0017XXXXXX number the call send Opensips to Asterisk



Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected
[4-2/1] <sip:a2b-asterisk-ip:5062>
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS
lookup...
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request:
sending:#012INVITE sip:0017xxxxx...@opensips-ip:5060
SIP/2.0#015#012Record-Route: <sip:opensips-ip;lr=on>#015#012Via: SIP/2.0/UDP
opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP
ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
4720779942 <sip:4720779...@opensips-ip:5060>;tag=1966722825#015#012To:
0017325824631 <sip:0017xxxx...@opensips-ip:5060>#015#012Call-ID:
32167199575863-11502744529...@ip-phoneip#015#012cseq: 2
INVITE#015#012Contact:
<sip:4720779...@ipphone-ip:5060>#015#012Proxy-Authorization:
Digest username="4720779942", realm="asterisk", nonce="79ee65ba",
uri="sip:0017xxx...@opensips-ip:5060",
response="3e182f165a5663d0b145d6b55d34e94b",
algorithm=MD5#015#012Max-Forwards: 69#015#012Supported:
replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK,
OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK,
UPDATE#015#012Content-Type: application/sdp#015#012Content-Length:
319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4
202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0
0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9
G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012.
opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220,
proto=1



when i ngrep
------------


U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060
INVITE sip:0017xxxxx...@asterisk-a2b-ip:5060 SIP/2.0.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
From: 4720779942 <sip:4720779...@asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017xxxxx...@asterisk-a2b-ip:5060>.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Contact: <sip:4720779...@ipphone:5060>.
Proxy-Authorization: Digest username="4720779942", realm="asterisk",
nonce="07ba8624", uri="sip:0017xxxxx...@asterisk-a2b-ip:5060",
response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5.
Max-Forwards: 70.
Supported: replaces.
User-Agent: Voip Phone 1.0.
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE.
Content-Type: application/sdp.
Content-Length: 319.
.
v=0.
o=4720779942 69102627 18481147 IN IP4 ipphone.
s=A conversation.
c=IN IP4 ipphone.
t=0 0.
m=audio 10034 RTP/AVP 18 4 8 0 9 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
From: 4720779942 <sip:4720779...@asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017xxxxx...@asterisk-a2b-ip:5060>.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Server: OpenSIPS (1.5.1-notls (i386/linux)).
Content-Length: 0.
.


U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060.
From: 4720779942 <sip:4720779...@asterisk-a2b-ip:5060>;tag=3037030266.
To: 0017XXXXXXXX <sip:0017xxxxx...@asterisk-a2b-ip:5060>;tag=as0cb075c5.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="07ba8624".
Content-Length: 0.

------

when i enable debug at Asterisk and Look at i see the below error
---------------------------------------------------------------

<--- SIP read from a2b-asterisk-ip:5060 --->
INVITE sip:0017xxxxxx...@a2b-asterisk-ip:5060 SIP/2.0
Record-Route: <sip:a2b-asterisk-ip;lr=on>
Via: SIP/2.0/UDP a2b-asterisk-ip;branch=z9hG4bK166.1b7e2827.0
Via: SIP/2.0/UDP
Ip-phone:5060;received=Ip-phone;branch=z9hG4bK295731884823024293;rport=5060
From: 4720779942 <sip:4720779...@a2b-asterisk-ip:5060>;tag=12544334
To: 0017XXXXXXXXX <sip:0017xxxxxx...@a2b-asterisk-ip:5060>
Call-ID: 16946271051109-143302828620...@ip-phone
CSeq: 1 INVITE
Contact: <sip:4720779...@ip-phone:5060>
Max-Forwards: 69
Supported: replaces
User-Agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 319

v=0
o=4720779942 31008195 22123120 IN IP4 Ip-phone
s=A conversation
c=IN IP4 Ip-phone
t=0 0
m=audio 10030 RTP/AVP 18 4 8 0 9 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
[Jun 30 01:15:29] VERBOSE[24612] logger.c: --- (15 headers 14 lines) ---
[Jun 30 01:15:29] VERBOSE[24612] logger.c: Ignoring this INVITE request
[Jun 30 01:15:31] VERBOSE[24612] logger.c: Reliably Transmitting (no NAT) to
termination-provider-ip:5062:
OPTIONS sip:termination-provider-ip:5062 SIP/2.0
Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport
From: "asterisk" <sip:aster...@a2b-asterisk-ip:5062>;tag=as4cf91fd8
To: <sip:termination-provider-ip:5062>
Contact: <sip:aster...@a2b-asterisk-ip:5062>
Call-ID: 65a49c0977c6de0a1d2dbbfe75772...@a2b-asterisk-ip
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Jun 2009 08:15:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
[Jun 30 01:15:32] VERBOSE[24612] logger.c:
<--- SIP read from termination-provider-ip:5062 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport=5062
From: "asterisk" <sip:aster...@a2b-asterisk-ip:5062>;tag=as4cf91fd8
To:
<sip:termination-provider-ip:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.ee5a
Call-ID: 65a49c0977c6de0a1d2dbbfe75772...@a2b-asterisk-ip
CSeq: 102 OPTIONS
Content-Length: 0

---------


why does Asterisk sending with out any values

---

From: "asterisk" <sip:aster...@a2b-asterisk-ip:5062>;tag=as4cf91fd8
To: <sip:termination-provider-ip:5062>

---

Any suggestions

Ram
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